/external/zlib/src/ |
H A D | make_vms.com | 706 $! Write configuration to both permanent and temporary config file
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/external/webrtc/webrtc/test/ |
H A D | rtp_rtcp_observer.h | 85 const FakeNetworkPipe::Config& configuration) 86 : test::DirectTransport(configuration, send_call), 82 PacketTransport(Call* send_call, RtpRtcpObserver* observer, TransportType transport_type, const FakeNetworkPipe::Config& configuration) argument
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/external/webrtc/webrtc/video/ |
H A D | vie_channel.cc | 1110 RtpRtcp::Configuration configuration; local 1111 ReceiveStatistics* null_receive_statistics = configuration.receive_statistics; 1112 configuration.audio = false; 1113 configuration.receiver_only = receiver_only; 1114 configuration.receive_statistics = receive_statistics; 1115 configuration.outgoing_transport = outgoing_transport; 1116 configuration.intra_frame_callback = intra_frame_callback; 1117 configuration.rtt_stats = rtt_stats; 1118 configuration.rtcp_packet_type_counter_observer = 1120 configuration [all...] |
/external/webrtc/webrtc/voice_engine/ |
H A D | channel.cc | 915 RtpRtcp::Configuration configuration; local 916 configuration.audio = true; 917 configuration.outgoing_transport = this; 918 configuration.audio_messages = this; 919 configuration.receive_statistics = rtp_receive_statistics_.get(); 920 configuration.bandwidth_callback = rtcp_observer_.get(); 921 configuration.paced_sender = rtp_packet_sender_proxy_.get(); 922 configuration.transport_sequence_number_allocator = 924 configuration.transport_feedback_callback = feedback_observer_proxy_.get(); 926 _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration)); [all...] |
/external/webrtc/webrtc/modules/desktop_capture/ |
H A D | mouse_cursor_monitor_mac.mm | 106 MacDesktopConfiguration configuration = 113 for (MacDisplayConfigurations::iterator it = configuration.displays.begin(); 114 it != configuration.displays.end(); ++it) { 206 configuration.FindDisplayConfigurationById( 218 position.subtract(configuration.bounds.top_left());
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
H A D | rtp_rtcp.h | 81 * configuration - Configuration of the RTP/RTCP module. 83 static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); 130 * A combination of the configuration MaxTransferUnit and 140 * A combination of the configuration MaxTransferUnit, headers and
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | nack_rtx_unittest.cc | 179 RtpRtcp::Configuration configuration; variable 180 configuration.audio = false; 181 configuration.clock = &fake_clock; 183 configuration.receive_statistics = receive_statistics_.get(); 184 configuration.outgoing_transport = &transport_; 185 rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
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H A D | rtcp_format_remb_unittest.cc | 87 RtpRtcp::Configuration configuration; local 88 configuration.audio = false; 89 configuration.clock = system_clock_; 90 configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get(); 91 configuration.outgoing_transport = &null_transport_; 92 dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
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H A D | rtcp_receiver_unittest.cc | 80 RtpRtcp::Configuration configuration; local 81 configuration.audio = false; 82 configuration.clock = &system_clock_; 83 configuration.outgoing_transport = test_transport_; 84 configuration.remote_bitrate_estimator = remote_bitrate_estimator_.get(); 85 rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(configuration);
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H A D | rtcp_sender_unittest.cc | 229 RtpRtcp::Configuration configuration; local 230 configuration.audio = false; 231 configuration.clock = &clock_; 232 configuration.outgoing_transport = &test_transport_; 234 rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
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H A D | rtp_rtcp_impl.cc | 48 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) { argument 49 if (configuration.clock) { 50 return new ModuleRtpRtcpImpl(configuration); 54 memcpy(&configuration_copy, &configuration, 61 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) argument 62 : rtp_sender_(configuration.audio, 63 configuration.clock, 64 configuration.outgoing_transport, 65 configuration.audio_messages, 66 configuration [all...] |
H A D | rtp_rtcp_impl.h | 31 explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
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/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api.cc | 92 RtpRtcp::Configuration configuration; variable 93 configuration.audio = true; 94 configuration.clock = &fake_clock_; 95 configuration.outgoing_transport = &null_transport_; 96 module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
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H A D | test_api_audio.cc | 104 RtpRtcp::Configuration configuration; variable 105 configuration.audio = true; 106 configuration.clock = &fake_clock; 107 configuration.receive_statistics = receive_statistics1_.get(); 108 configuration.outgoing_transport = transport1; 109 configuration.audio_messages = audioFeedback; 111 module1 = RtpRtcp::CreateRtpRtcp(configuration); 116 configuration.receive_statistics = receive_statistics2_.get(); 117 configuration.outgoing_transport = transport2; 118 configuration [all...] |
H A D | test_api_rtcp.cc | 87 RtpRtcp::Configuration configuration; local 88 configuration.audio = true; 89 configuration.clock = &fake_clock; 90 configuration.receive_statistics = receive_statistics1_.get(); 91 configuration.outgoing_transport = transport1; 92 configuration.intra_frame_callback = myRTCPFeedback1; 99 module1 = RtpRtcp::CreateRtpRtcp(configuration); 107 configuration.receive_statistics = receive_statistics2_.get(); 108 configuration.outgoing_transport = transport2; 109 configuration [all...] |
H A D | test_api_video.cc | 47 RtpRtcp::Configuration configuration; local 48 configuration.audio = false; 49 configuration.clock = &fake_clock; 50 configuration.outgoing_transport = transport_; 52 video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
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/external/webrtc/webrtc/modules/video_coding/test/ |
H A D | rtp_player.cc | 220 RtpRtcp::Configuration configuration; local 221 configuration.clock = clock; 222 configuration.audio = false; 224 configuration.clock, handler->payload_sink_.get(), NULL,
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/external/webrtc/webrtc/modules/audio_device/android/ |
H A D | opensles_common.cc | 22 SLDataFormat_PCM configuration; local 23 configuration.formatType = SL_DATAFORMAT_PCM; 24 configuration.numChannels = kNumChannels; 29 configuration.samplesPerSec = sample_rate * 1000; 30 configuration.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; 31 configuration.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16; 32 configuration.channelMask = SL_SPEAKER_FRONT_CENTER; 33 if (2 == configuration.numChannels) { 34 configuration.channelMask = 37 configuration [all...] |
/external/webrtc/tools/python_charts/webrtc/ |
H A D | data_helper.py | 140 configurations: List of one ore more configurations. Each configuration 143 Example of a single configuration: 151 - a dictionary describing the columns in the configuration table to be 159 - a list containing dictionaries (one per configuration) with the 160 configuration column names mapped to the value for each test run: 175 for configuration in configurations: 178 for values in configuration:
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H A D | main.py | 23 - test_configuration - a dictionary of test configuration names and values. 92 messages.append('Invalid input file: %s. Missing configuration key ' 123 # (uses test configuration from the last data set, assuming it is the same 127 ShowErrorPage('Cannot configuration field named "bit_rate_in_kbps"') 144 def FindConfiguration(configuration, name): 145 """ Finds a configuration value using it's name. 146 Returns the first configuration with a matching name. Returns None if no 147 matching configuration is found. """ 149 for row in configuration:
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/external/webrtc/webrtc/common_audio/signal_processing/ |
H A D | Android.mk | 67 # Some new .s files have compilation error with AOSP configuration,
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/external/webrtc/talk/app/webrtc/objc/ |
H A D | RTCPeerConnection.mm | 211 - (BOOL)setConfiguration:(RTCConfiguration *)configuration { 213 configuration.nativeConfiguration);
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H A D | RTCPeerConnectionFactory.mm | 92 - (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration 96 config:configuration.nativeConfiguration
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/external/webrtc/talk/app/webrtc/objc/public/ |
H A D | RTCPeerConnection.h | 101 // Sets the PeerConnection's global configuration to |configuration|. 106 - (BOOL)setConfiguration:(RTCConfiguration *)configuration;
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H A D | RTCPeerConnectionFactory.h | 57 - (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
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